Method and appartus for transmitting coded audio signals through a transmission channel with limited bandwidth

ABSTRACT

A digital audio transmitter system capable of transmitting high quality, wideband speech over a transmission channel with a limited bandwidth such as a traditional telephone line. The digital audio transmitter system includes a coder for coding an input audio signal to a digital signal having a transmission rate that does not exceed the maximum allowable transmission rate for traditional telephone lines and a decoder for decoding the digital signal to provide an output audio signal with an audio bandwidth of wideband speech. A coder and a decoder may be provided in a single device to allow two-way communication between multiple devices.

A microfiche appendix, containing 3 sheets having a total of 188 frames,is included for all purposes.

FIELD OF THE INVENTION

The present invention relates generally to an apparatus and method fortransmitting audio signals and pertains, more specifically, to anapparatus and method for transmitting a high quality audio signal, suchas wideband speech, through a transmission channel having a limitedbandwidth or transmission rate.

BACKGROUND OF THE INVENTION

Human speech lies in the frequency range of approximately 7 Hz to 10kHz. Because traditional telephone systems only provide for thetransmission of analog audio signals in the range of about 300 Hz to3400 Hz or a bandwidth of about 3 kHz (narrowband speech), certaincharacteristics of a speaker's voice are lost and the voice soundssomewhat muffled. A telephone system capable of transmitting an audiosignal approaching the quality of face-to-face speech requires abandwidth of about 6 kHz (wideband speech).

Known digital transmission systems are capable of transmitting widebandspeech audio signals. However, in order to produce an output audiosignal of acceptable quality with a bandwidth of 6 kHz, these digitalsystems require a transmission channel with a transmission rate thatexceeds the capacity of traditional telephone lines. A digital systemtransmits audio signals by coding an input audio signal into a digitalsignal made up of a sequence of binary numbers or bits, transmitting thedigital signal through a transmission channel, and decoding the digitalsignal to produce an output audio signal. During the coding process thedigital signal is reduced or compressed to minimize the necessarytransmission rate of the signal. One known method for compressingwideband speech is disclosed in Recommendation G.722 (CCITT, 1988). Asystem using the compression method described in G.722 still requires atransmission rate of at least 48 kbit/s to produce wideband speech of anacceptable quality.

Because the maximum transmission rate over traditional telephone linesis 28.8 kbit/s using the most advanced modem technology, alternativetransmission channels such as satellite or fiber optics would have to beused with an audio transmission system employing the data compressionmethod disclosed in G.722. Use of these alternative transmissionchannels is both expensive and inconvenient due to their limitedavailability. While fiber optic lines are available, traditional coppertelephone lines now account for an overwhelming majority of existinglines and it is unlikely that this balance will change anytime in thenear future. A digital phone system capable of transmitting widebandspeech over existing transmission rate limited telephone phone lines istherefore highly desirable.

OBJECTS OF THE INVENTION

The disclosed invention has various embodiments that achieve one or moreof the following features or objects:

An object of the present invention is to provide for the transmission ofhigh quality wideband speech over existing telephone networks.

A further object of the present invention is to provide for thetransmission of high quality audio signals in the range of 20 Hz to atleast 5,500 Hz over existing telephone networks.

A still further object of the present invention is to accomplish datacompression on wideband speech signals to produce a transmission rate of28.8 kbit/s or less without significant loss of audio quality.

A still further object of the present invention is to provide a devicewhich allows a user to transmit and receive high quality wideband speechand audio over existing telephone networks.

A still further object of the present invention is to provide a portabledevice which is convenient to use and allows ease of connection toexisting telephone networks.

A still further object of the present invention is to provide a devicewhich is economical to manufacture.

A still further object of the present invention is to provide easy andflexible programmability.

SUMMARY OF THE INVENTION

In accordance with the present invention, the disadvantages of the priorart have been overcome by providing a digital audio transmitter systemcapable of transmitting high quality, wideband speech over atransmission channel with a limited bandwidth such as a traditionaltelephone line.

More particularly, the digital audio transmitter system of the presentinvention includes a coder for coding an input audio signal to a digitalsignal having a transmission rate that does not exceed the maximumallowable transmission rate for traditional telephone lines and adecoder for decoding the digital signal to provide an output audiosignal with an audio bandwidth of wideband speech. A coder and a decodermay be provided in a single device to allow two-way communicationbetween multiple devices. A device containing a coder and a decoder iscommonly referred to as a CODEC (COder/DECoder).

These and other objects, advantages and novel features of the presentinvention, as well as details of an illustrative embodiment thereof,will be more fully understood from the following description and fromthe drawings.

BRIEF DESCRIPTION OF THE DRAWINGS

FIG. 1 is a block diagram of a digital audio transmission systemincluding a first CODEC and second CODEC in accordance with the presentinvention.

FIG. 2 is a block diagram of a CODEC of FIG. 1.

FIG. 3 is a block diagram of an audio input/output circuit of a CODEC.

FIG. 4 is a detailed circuit diagram of the audio input portion of FIG.3.

FIG. 5 is a detailed circuit diagram of the level LED's portion of FIG.3.

FIG. 6 is a detailed circuit diagram of the headphone amp portion ofFIG. 3.

FIG. 7 is a block diagram of a control processor of a CODEC.

FIG. 8 is a detailed circuit diagram of the microprocessor portion ofthe control processor of FIG. 7.

FIG. 9 is a detailed circuit diagram of the memory portion of thecontrol processor of FIG. 7.

FIG. 10 is a detailed circuit diagram of the dual UART portion of thecontrol processor of FIG. 7.

FIG. 11 is a detailed circuit diagram of the keypad, LCD display andinterface portions of the control processor of FIG. 7.

FIG. 12 is a block diagram of an encoder of a CODEC.

FIG. 13 is a detailed circuit diagram of the encoder digital signalprocessor and memory portions of the encoder of FIG. 12.

FIG. 14 is a detailed circuit diagram of the clock generator portion ofthe encoder of FIG. 12.

FIG. 15 is a detailed circuit diagram of the Reed-Soloman encoder anddecoder portions of FIGS. 12 and 16.

FIG. 16 is a block diagram of a decoder of a CODEC.

FIG. 17 is a detailed circuit diagram of the encoder digital signalprocessor and memory portions of the decoder of FIG. 16.

FIG. 18 is a detailed circuit diagram of the clock generator portion ofthe decoder of FIG. 16.

FIG. 19 is a detailed circuit diagram of the analog/digital converterportion of the encoder of FIG. 12.

FIG. 20 is a detailed circuit diagram of the digital/analog converterportion of the decoder of FIG. 16.

DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENT

A digital audio transmission system 10, as shown in FIG. 1, includes afirst CODEC (COder/DECoder) 12 for transmitting and receiving a widebandaudio signal such as wideband speech to and from a second CODEC 14 via atraditional copper telephone line 16 and telephone network 17. Whentransmitting an audio signal, the first CODEC 12 performs a codingprocess on the input analog audio signal which includes converting theinput audio signal to a digital signal and compressing the digitalsignal to a transmission rate of 28.8 kbit/s or less. The preferredembodiment compresses the digital signal using a modified version of theISO/MPEG (International Standards Organization/Motion Picture ExpertGroups) compression scheme according to the software routine disclosedin the microfiche software appendix filed herewith. The coded digitalsignal is sent using standard modem technology via the telephone line 16and telephone network 17 to the second CODEC 14. The second CODEC 14performs a decoding process on the coded digital signal by correctingtransmission errors, decompressing the digital signal and reconvertingit to produce an output analog audio signal.

FIG. 2 shows a CODEC 12 which includes an analog mixer 20 for receiving,amplifying, and mixing an input audio signal through a number of inputlines. The input lines may include a MIC line 22 for receiving an analogaudio signal from a microphone and a generic LINE 24 input for receivingan analog audio signal from an audio playback device such as a tapedeck. The voltage level of an input audio signal on either the MIC line22 or the generic LINE 24 can be adjusted by a user of the CODEC 12 byadjusting the volume controls 26 and 28. When the analog mixer 20 isreceiving an input signal through both the MIC line 22 and the genericLINE 24, the two signals will be mixed or combined to produce a singleanalog signal. Audio level LED's 30 respond to the voltage level of amixed audio signal to indicate when the voltage exceeds a desiredthreshold level. A more detailed description of the analog mixer 20 andaudio level LED's 30 appears below with respect to FIGS. 3 and 4.

The combined analog signal from the analog mixer 20 is sent to theencoder 32 where the analog signal is first converted to a digitalsignal. The sampling rate used for the analog to digital conversion ispreferably one-half the transmission rate of the signal which willultimately be transmitted to the second CODEC 14 (shown in FIG. 1).After analog to digital conversion, the digital signal is thencompressed using a modified version of the ISO/MPEG algorithm. TheISO/MPEG compression algorithm is modified to produce a transmissionrate of 28.8 kbit/s. This is accomplished by the software routine thatis disclosed in the software appendix.

The compressed digital signal from the encoder 32 is then sent to anerror protection processor 34 where additional error protection data isadded to the digital signal. A Reed-Solomon error protection format isused by the error protection processor 34 to provide both burst andrandom error protection. The error protection processor 34 is describedbelow in greater detail with respect to FIGS. 12 and 15.

The compressed and error protected digital signal is then sent to ananalog modem 36 where the digital signal is converted back to an analogsignal for transmitting. As shown in FIG. 1, this analog signal is sentvia a standard copper telephone line 16 through a telephone network 17to the second CODEC 14. The analog modem 36 is preferably a V.34synchronous modem. This type of modem is commercially available.

The analog modem 36 is also adapted to receive an incoming analog signalfrom the second CODEC 14 (or another CODEC) and reconvert the analogsignal to a digital signal. This digital signal is then sent to an errorcorrection processor 38 where error correction according to aReed-Soloman format is performed.

The corrected digital signal is then sent to a decoder 40 where it isdecompressed using the modified version of the ISO/MPEG algorithm asdisclosed in the software appendix. After decompression the digitalsignal is converted to an analog audio signal. A more detaileddescription of the decoder 40 appears below with respect to FIGS. 7, 16,17 and 18. The analog audio signal may then be perceived by a user ofthe CODEC 12 by routing the analog audio signal through a headphone amp42 wherein the signal is amplified. The volume of the audio signal atthe headphone output line 44 is controlled by volume control 46.

The CODEC 12 includes a control processor 48 for controlling the variousfunctions of the CODEC 12 according to software routines stored inmemory 50. A more detailed description of the structure of the controlprocessor appears below with respect to FIGS. 7, 8, 9, 10, and 11. Onesoftware routine executed by the control processor allows the user ofthe CODEC 12 to initiate calls and enter data such as phone numbers.When a call is initiated the control processor sends a signal includingthe phone number to be dialed to the analog modem 36. Data entry isaccomplished via a keypad 52 and the entered data may be monitored byobservation of an LCD 54. The keypad 52 also includes keys for selectingvarious modes of operation of the CODEC 12. For example, a user mayselect a test mode wherein the control processor 48 controls the signalpath of the output of the encoder to input of decoder to bypass thetelephone network allows testing of compression and decompressionalgorithms and their related hardware. Also stored in memory 50 is thecompression algorithm executed by the encoder 32 and the decompressionalgorithm executed by the decoder 40.

Additional LED's 56 are controlled by the control processor 48 and mayindicate to the user information such as "bit synchronization" (achievedby the decoder) or "power on". An external battery pack 58 is connectedto the CODEC 12 for supplying power.

FIG. 3 shows a lower level block diagram of the analog mixer 20, audiolevel LED's 30 and analog headphone amp 42 as shown in FIG. 2. FIGS. 4,5 and 6 are the detailed circuit diagrams corresponding to FIG. 3.

Referring to FIG. 3 and 4, line input 210 is an incoming line levelinput signal while mic input 220 is the microphone level input. Thesesignals are amplified by a line amp 300 and a mic amp 302 respectivelyand their levels are adjusted by line level control 304 and mic levelcontrol 306 respectively. The microphone and line level inputs are fedto the input mixer 308 where they are mixed and the resulting combinedaudio input signal 310 is developed.

Referring now to FIGS. 3 and 5, the audio input signal 310 is sent tothe normal and overload signal detectors, 312 and 314 respectively,where their level is compared to a normal threshold 316 which defines anormal volume level and a clip threshold 318 which defines an overloadvolume level. When the audio input signal 310 is at a normal volumelevel a NORM LED 320 is lighted. When the audio input signal 310 is atan overload volume level a CLIP LED 322 is lighted.

Referring now to FIGS. 3 and 6, the audio input signal 310 is fed intothe record monitor level control 324, where its level is adjusted beforebeing mixed with the audio output signal 336 from the digital/analogconverter 442 (shown in FIGS. 16 and 20). The audio output signal 336 isfed to the local monitor level control 326 before it is fed into theheadphone mixer amplifier 334. The resulting output signal from theheadphone mixer amplifier 334 goes to a headphone output connector 338on the exterior of the CODEC 12 where a pair of headphones may beconnected.

The audio input signal 310 and audio output signal 336 are fed to recordmix control 328 which is operable by the user. The output of thiscontrol is fed to a mix level control 330 (also operable by a user) andthen to the record output amplifier 332. The resulting output signal ofthe record output amplifier 332 goes to a record output 340 on theexterior of the CODEC 12.

FIG. 7 shows a lower level block diagram of the control processor 48(shown in FIG. 2). The encoder 406 (referenced as number 32 in FIG. 2)is further described in FIG. 12 while the decoder 416 (referenced asnumber 40 in FIG. 2) is refined in FIG. 16. FIGS. 8, 9, 10, 11, 13, 14,15, 17, 18, 19 and 20 are detailed circuit diagrams.

Referring to FIGS. 7 and 8 the microprocessor 400 is responsible for thecommunication between the user, via keypad 412 and LCD display 414, andthe CODEC 12. The keypad 412 is used to input commands to the systemwhile the LCD display 414, is used to display the responses of thekeypad 412 commands as well as alert messages generated by the CODEC 12.

Referring now to FIGS. 7 and 9, the RAM (random access memory) 402 isused to hold a portion of the control processor control softwareroutines. The flash ROM (read only memory) 404 holds the softwareroutine (disclosed in the software appendix) which controls the modifiedISO/MPEG compression scheme performed by encoder DSP 406 and themodified ISO/MPEG decompression scheme performed by the decoder DSP 416,as well as the remainder of the control processor control softwareroutines.

Referring now to FIGS. 7 and 10, the dual UART (universal asynchronousreceiver/transmitter) 408 is used to provide asynchronous input/outputfor the control processor 48. The rear panel remote control port 409 andthe rear panel RS232 port 411 are used to allow control by an externalcomputer. This external control can be used in conjunction with orinstead of the keypad 412 and/or LCD display 414.

Referring now to FIGS. 7 and 11, the programmable interval timer circuit410 is used to interface the control processor with the keypad and LCDdisplay.

Referring now to FIGS. 7, 8 and 13, the encoder DSP (digital signalprocessor) 434 receives a digital pulse code modulated signal 430 fromthe analog/digital converter 450. The encoder DSP 434 performs themodified ISO/MPEG compression scheme according to the software routine(described in the software appendix) stored in RAM memory 436 to producea digital output 418.

The A/D clock generation unit 439 is shown in FIGS. 12 and 14. Thefunction of this circuitry is to provide all the necessary timingsignals for the analog digital converter 450 and the encoder DSP 434.

The Reed-Soloman error correction encoding circuitry 438 is shown inFIGS. 12 and 15. The function of this unit is to add parity informationto be used by the Reed-Soloman decoder 446 (also shown in FIG. 16) torepair any corrupted bits received by the Reed-Soloman decoder 446. TheReed-Soloman corrector 438 utilizes a shortened Reed-Soloman GF(256)code which might contain, for example, code blocks containing 170eight-bit data words and 8 eight-bit parity words.

Referring now to FIGS. 7, 16 and 17, the decoder DSP 440 receives adigital input signal 422 from the modem 36 (shown in FIG. 2). Thedecoder DSP 440 performs the modified ISO/MPEG decompression schemeaccording to the software routine (described in the software appendix)stored in RAM memory 444 to produce a digital output to be sent to thedigital/analog converter 422.

The D/A clock generation unit 448 is shown in FIGS. 16 and 18. Thefunction of this circuitry is to provide all the necessary timingsignals for the digital/analog converter 442 and the decoder DSP 440.

The analog/digital converter 450, shown in FIGS. 12 and 19, is used toconvert the analog input signal 310 into a PCM digital signal 430.

The digital/analog converter 442, shown in FIGS. 16 and 20 is used toconvert the PCM digital signal from the decoder DSP 440 into an analogaudio output signal 336.

The Reed-Soloman error correction decoding circuitry 446, shown in FIGS.15 and 16, decodes a Reed-Soloman coded signal to correct errorsproduced during transmission of the signal through the modem 36 (shownin FIG. 2) and telephone network.

Another function contemplated by this invention is to allow real time,user operated adjustment of a number of psycho-acoustic parameters ofthe ISO/MPEG compression/decompression scheme used by the CODEC 12. Amanner of implementing this function is described in applicant'sapplication entitled "System For Adjusting Psycho-Acoustic Parameters InA Digital Audio Codec" which is being filed concurrently herewith (suchapplication and related Software Appendix are hereby incorporated byreference). Also, applicants application entitled "System ForCompression And Decompression Of Audio Signals For Digital Transmission"and related Software Appendix which are being filed concurrentlyherewith are hereby incorporated by reference.

This invention has been described above with reference to a preferredembodiment. Modifications and variations may become apparent to oneskilled in the art upon reading and understanding this specification. Itis intended to include all such modifications and alterations within thescope of the appended claims.

What is claimed is:
 1. A portable CODEC for transmitting high qualityaudio signals over a standard telephone line having a limited bandwidthand a maximum transmission rate, said portable CODEC comprising:a singleportable housing; an analog mixer, within the housing, receiving aninput audio signal from at least one input line, said audio mixeramplifying and mixing input audio signals to produce a single combinedaudio input signal; memory, within the housing, storing a lossycompression routine and storing at least one set of predefinedpsycho-acoustic parameter levels for said compression routine; anencoder, within the housing, converting said single combined audio inputsignal to a combined digital input signal at a predefined sampling rateand encoding said combined digital input signal based on said lossycompression routine stored in memory to produce a single encoded digitalsignal having a predefined compression ratio with respect to said singlecombined audio input signal; an analog modem, with the housing,establishing a connection with, and a transmission rate for, a standardtelephone line of a telephone network, said modem converting saidencoded digital signal to an encoded analog output signal and outputtingsaid encoded analog output signal at said transmission rate establishedby said analog modem along the standard telephone line through thetelephone network; and a control processor, within the housing, definessaid predefined sampling rate for said encoder based on saidtransmission rate established by said analog modem to enable said analogmodem to output said encoded analog output signal at a transmission ratethat does not exceed the maximum transmission rate of the telephoneline.
 2. A portable CODEC according to claim 1, further comprising aclock generator providing synchronous clock signals to said encoder andanalog modem.
 3. A portable CODEC according to claim 1, wherein saidcontroller defines said sampling rate to equal approximately one-half ofsaid transmission rate established by said analog modem.
 4. A portableCODEC according to claim 1, further comprising:a microphone connected toa microphone input line, said microphone input line receiving live, realtime analog audio signals.
 5. A portable CODEC according to claim 1,further comprising:an input line adapted to receive an analog audiosignal from an audio playback device.
 6. A portable CODEC according toclaim 1, further comprising means for adjusting a voltage level of inputaudio signals on said at least one input line.
 7. A portable CODECaccording to claim 1, wherein said analog mixer receives, amplifies andmixes at least two input audio signals to produce said single combinedaudio input signal.
 8. A portable CODEC according to claim 1, furthercomprising audio level LEDs connected to said audio mixer indicatingwhen a voltage level of said single combined audio input signal exceedsa threshold level.
 9. A portable CODEC according to claim 1, whereinsaid analog mixer comprises:line amplifiers amplifying input audiosignals on at least two input lines; line level controllers, connectedto said amplifiers, adjustable by a user, said level controllerscontrolling an output voltage to which input audio signals are amplifiedby said amplifiers; and an input mixer mixing amplified audio signalsoutput by said level controllers to produce said single combined audioinput signal.
 10. A portable CODEC according to claim 1, wherein saidanalog mixer comprises:normal and overload signal detectors comparingsaid single combined audio input signal with normal and clip thresholdsdefining normal and overload volume levels, respectively; and normal andoverload LEDs connected to said normal and overload signal detectors,respectively, said normal LED lighting when said single combined audioinput signal is at said normal threshold, said overload LED lightingwhen said single combined audio input signal is at said overloadthreshold.
 11. A portable CODEC according to claim 1, wherein saidencoder encodes said combined digital input signal based on predefinedpsycho-acoustic parameter levels stored in memory that produce encodeddigital signals having a bandwidth range of approximately 20 Hz to 5,500Hz.
 12. A portable CODEC according to claim 1, wherein said encoderencodes said combined digital input signal based on predefinedpsycho-acoustic parameter levels stored in memory that produce encodeddigital signals having a bandwidth range of approximately 300 Hz to3,000 Hz.
 13. A portable CODEC according to claim 1, wherein saidencoder encodes said combined digital input signal based on an ISO/MPEGlayer II compression routine having predefined psycho-acoustic parameterlevels that produce an encoded digital signal having a bandwidth rangeof approximately 20 Hz to 5,500 Hz.
 14. A portable CODEC according toclaim 1, further comprising:an error protection processor adding errorprotection data to said single encoded digital signal based on apredefined error protection format to produce an encoded and errorprotected digital signal, said analog modem outputting said encoded anderror protected digital signal as said output signal.
 15. A portableCODEC according to claim 14, wherein said predefined error protectionformat is a Reed-Solomon error protection format, said error protectionprocessor providing both burst and random error protection.
 16. Aportable CODEC according to claim 1, wherein said analog modem receivesa single incoming encoded analog signal from said standard telephoneline on said telephone network, said modem converting said singleincoming encoded analog signal to a single incoming encoded digitalsignal.
 17. A portable CODEC according to claim 16, wherein saidincoming encoded analog signal contains error protection data, saidCODEC further comprising:an error correction processor performing errorcorrection upon said incoming encoded digital signal based on said errorprotection data to produce an incoming error corrected encoded digitalsignal.
 18. A portable CODEC according to claim 17, wherein said errorcorrection processor comprises:an error correction encoding circuitgenerating parity information based on said incoming encoded digitalsignal; and a Reed-Solomon encoder receiving and preparing corrupteddata bits in said incoming encoded digital signal based on said parityinformation to correct errors produced during transmission through thetelephone network.
 19. A portable CODEC according to claim 18, wherein acode of said Reed-Solomon encoder includes code blocks containingapproximately 178-bit data words and 8-bit parity words.
 20. A portableCODEC according to claim 16, further comprising:a decoder decoding saidincoming encoded digital signal from said analog modem based on a lossydecompression routine stored in memory to provide an analog outputsignal.
 21. A portable CODEC according to claim 20, wherein said controlprocessor is selectable by a user between multiple modes of operation,said control processor, when in a test mode, bypassing said telephonenetwork and directing said single encoded digital signal from saidencoder directly to said decoder to allow testing of said compressionand decompression routines stored in memory.
 22. A portable CODECaccording to claim 20, further comprising a clock generator forproviding synchronized clock signals to said encoder and decoder.
 23. Aportable CODEC according to claim 20, wherein said decodercomprises:memory storing an ISO/MPEG decompression routine; and adigital single processor decoding and converting said incoming encodeddigital signal based on said ISO/MPEG decompression routine stored inmemory to produce said analog output signal.
 24. A portable CODECaccording to claim 23, wherein said decoder further comprises:a D/Aconverter converting a digital output of said digital signal processorto said analog telephone signal.
 25. A portable CODEC according to claim24, wherein said decoder further comprises a D/A clock generation unitgenerating synchronous timing signals for said D/A converter and digitalsignal processor.
 26. A portable CODEC according to claim 1, furthercomprising:a headphone amplifier outputting said analog output signal toa headphone output line; and a volume control controlling a volume ofsaid analog output signal at said telephone output line.
 27. A portableCODEC according to claim 26, wherein said headphone amplifier furthercomprises:record and local monitor level controls receiving andadjusting levels of said single combined audio input signal from saidanalog mixer and of said analog output signal from said decoder,respectively; and a headphone mixer amplifier mixing output signals ofsaid record and local monitored level controls to output a mixedrecord/local output signal at said headphone output line.
 28. A portableCODEC according to claim 26, wherein said headphone amplifier furthercomprises:a record mix controller operative by the user, receiving saidcombined audio signal from said analog mixer, said mix controllercontrolling a level of said combined audio input signal; and a recordoutput amplifier controlled by said record mix controller outputtingsaid combined audio input signal at a desired level to a record output.29. A portable CODEC according to claim 1, wherein said controlprocessor comprises:a keypad/LCD interface adapted to communicate with akeypad and LCD display respectively; and a microprocessor communicatingwith the user through the keypad/LCD interface.
 30. A portable CODECaccording to claim 1, further comprising:a keypad entering inputcommands to said control processor; and a LCD display displayingresponses to said input commands and displaying alert messages.
 31. Aportable CODEC according to claim 30, further comprising:a programmableinterval timer circuit interfacing said control processor with saidkeypad and LCD display.
 32. A portable CODEC according to claim 31,further comprising:a universal asynchronous receiver/transmitterproviding a synchronous input/output data to said control processor froman external computer through a rear panel remote control port and rearRS 232 port in said receiver/transmitter.
 33. A portable CODEC accordingto claim 1, wherein said encoder comprises:an A/D converter convertingsaid combined audio input signal to a digital pulse code modulatedsignal at said predefined sampling rate; and a digital signal processorencoding said digital pulse code modulated signal based on a modifiedISO/MPEG compression routine stored in said memory to produce saidencoded signal.
 34. A portable CODEC according to claim 33, furthercomprising:an A/D clock generation unit generating timing signals forsaid A/D converter and digital signal processor based on saidtransmission rate established by said analog modem.